[asterisk-users] SIP_CODEC variable

[Olivier]

2009/2/25 Jeff LaCoursiere <jeff at jeff.net>

>
> On Wed, 25 Feb 2009, Jared Smith wrote:
>
> > On Wed, 2009-02-25 at 07:54 -0500, Mike wrote:
> >> I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the
> >> appropriate Page command call. But I get this in th CLI:
> >
> >> NOTICE[4764]: chan_sip.c:3706 try_suggested_sip_codec: Ignoring
> >> ${SIP_CODEC} variable because it is not shared by both ends.
> >
> > This is a wild guess (and I don't currently have the time to check it
> > out properly), but if my memory serves me the Polycom phones don't
> > support the GSM codec.  You might try ulaw instead.
> >
>
> True, that.  They do G.729 though!
>
> j


If my memory serves me right, there is an opened bug in Mantis about
SIP_CODEC not being presently applied to both legs of a call.


>
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